Re: [asterisk-users] Asterisk 11- Answer with [m=image 0 udptl t38] and Call Drop
On 1/18/13 13:24 , Matthew Jordan wrote: 1) Contact your carrier and ask why they are rejecting the 200 OK. 2) Assuming they won't change their behaviour, find out what they want in a response that declines an image media format. Without knowing what your carrier thinks the SDP should look like, any modifications you make to Asterisk will be guesses. I ran into a similar problem this week. There's a number of SIP implementations (either legacy or not good enough) that don't handle a zero port denial of a media stream quite right. In the past these implementations would have worked fine when the called party would have just ignored the offending media stream, instead of sending an explicit deny. -- Andreas Sikkema -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Complex Call Distribution
Hello, I have Elastix ISO install (FreePBX 2.7.0.3) My current Setup is as follows: Inbound Route Queue (Dynamic Agents) The queue distributes calls based on rrMemory. I have been asked to redesign the call distribution as follows: Calls will be delievered to Level-1 Agents (say 4 dynamic agents) in rrMemory format. When Level-1 Agents are busy, distribute calls to Level-2 Agents (say 3 dynamic agents) in rrMemory format. When Level-2 Agents are busy, distribute calls to Level-3 Agents (say 2 dynamic agents) in rrMemory format. Is it possible to setup the call distribution in the above format using any kind of logic or algorithm ? I tried using Penalties function in Queues. Created 2 penalties : 0 (level-1) and 1000 (level-2) and assigned penalties to agents (static) I made a few test calls, but Level-2 agents were delivered calls inspite of Level-1 agents being available. Any help or pointers are appreciated. Thx, Vai -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime vs Static Files
It is really unbelievable ... I was thinking: Asterisk uses an internal database to maintain states of peers. It is usually located in /var/lib/asterisk/astdb and it is a berkely db, but other database backends seem available. Are you sharing also this database between the two servers? It is the only option left... The only thing shared is the sip realtime db. I think i'm going to try removing the sip realtime db and automate the creation of the sip.conf file and issuing of the 'sip reload' and see if the problem goes away. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 11's app_page options
I have just upgraded to asterisk 11 from 1.8 I have noticed that my Page command: exten = 1,1,Page(SIP/101,diqA(local/intercom)) does not play the local/intercom sound to the conference. according to the doc at https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Page , it seems like it still should. is there something i need to do to make this work how i expect it? my confbridge.conf is vanilla; i dont see anything that needs changing. also, when the conference ends, the CLI shows: [Jan 25 23:50:52] ERROR[3746][C-000a]: confbridge/conf_state.c:47 conf_invalid_event_fn: Invalid event for confbridge user '' [Jan 25 23:50:52] ERROR[3745][C-000a]: confbridge/conf_state.c:47 conf_invalid_event_fn: Invalid event for confbridge user '' any way to hush/fix that? The Page application uses the ConfBridge application to implement its features. You have found two bugs in confbridge: 1) The CONFBRIDGE(user,announcement) file does not get played. The code to do that apparently got removed accidentally when confbridge was restructured to be more state machine like. 2) The error message is another problem with the state machine. Please create an issue in the issue tracker: https://issues.asterisk.org/jira Thanks. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 11's app_page options
On 1/26/2013 4:00 PM, Richard Mudgett wrote: features. You have found two bugs in confbridge: Issues created in jira. thanks for your input! -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime vs Static Files
It is a shame we were unable to find the solution to your problem. Do you want to setup a test system like the good one and let me access it to check what is going on? I am really really curious. Leandro Il giorno 26/gen/2013 19:49, Dan Journo d...@keshercommunications.com ha scritto: It is really unbelievable ... I was thinking: Asterisk uses an internal database to maintain states of peers. It is usually located in /var/lib/asterisk/astdb and it is a berkely db, but other database backends seem available. Are you sharing also this database between the two servers? It is the only option left... ** ** The only thing shared is the sip realtime db. ** ** I think i'm going to try removing the sip realtime db and automate the creation of the sip.conf file and issuing of the 'sip reload' and see if the problem goes away. ** ** ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
